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<title>TooLAME: MPEG 1/2 Layer II Audio Encoder</title>
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<h2>tooLAME - an optimized mpeg 1/2 layer 2 audio encoder</h2>
Copyright (C) 2002, 2003 Michael Cheng [mikecheng at NOT planckenergy com] remove the NOT
http://www.planckenergy.com/</p>

<h2>Contents</h2>
<UL>
<LI> LGPL 
<LI> Introduction
<LI> Usage
<LI> Examples
<LI> Contributors
<LI> References
</UL>

<h2>LGPL</h2>
<pre>
All changes to the ISO source are licensed under the LGPL  
(see LGPL.txt for details)

tooLAME is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.

tooLAME is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
Lesser General Public License for more details.

You should have received a copy of the GNU Lesser General Public
License along with tooLAME; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
</pre>

<h2>Introduction</h2>

tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder.  It is based heavily on 
<UL>
    <LI>the ISO dist10 code 
    <LI>improvement to algorithms as part of the LAME project (www.sulaco.org/mp3)
    <LI>work by myself and other contributors (see CONTRIBUTORS)
</ul>
<h2>Installation</h2>
<OL>
<LI> edit Makefile (at least change the architecture type [ARCH] to suit your machine)
<LI> 'make'
</ol>

<h2>Usage</h2>

<pre>
         ./toolame [options] [input file] [output file]

Input File
	tooLAME parses AIFF and WAV files for file info
	raw PCM is assumed if no header is found
	for stdin use a -

Output File
	file is automatically renamed from *.* to *.mp2
	for stdout use a -

Input Options
	-s [int]
		if inputting raw PCM sound, you must specify the sample rate
		default sample rate is 44.1khz.

	-a	
		downmix from stereo to mono
		if the incoming file is stereo, combine the audio into 
		a single channel

	-x 	
		force byte-swapping of the input.  (current endian detection is dodgy, 
		so if toolame produces only noise, use -x )

	-g
		swap the LR channels of a stereo file

Output Options
	-m [char]
		the encoding mode (default 'j')
		's' stereo
		'd' dual channel
		'j' joint stereo
		'm' mono

	-p [int]
		which psy model to use (default '1')
		Different models for the psychoacoustics
	        Models: -1 to 4

	-b [int]
		the total bitrate 	
		For 48/44.1/32kHz default = 192 
		For 24/22.05/16kHz default = 96

	-v [int]
		Switch on VBR mode.
		The higher the number the better the quality.
		Useful range -10 to 10.
		See README.VBR for details.
		
		
Operation
	-f 	
		fast mode turns off calculation of the psychoacoustic model.
		Instead a set of default values are assumed

	-q [int]
		quick mode calculates the psy model every 'num' frames.

Misc
	-d emp
		de-emphasis (default 'n')
	-c 	
		mark as copyright
	-o
		mark as original
	-e
		add error protection
	-r
		force padding bits off
	-D
		add DAB extensions
	-t [int]
		'talkativity' setting. 0 = no message. 3 = too much information
                (-t 20 will probably flood you off your terminal)
</pre>

<h2>Examples</h2>

<pre>
	toolame sound.wav
</pre>
	This will encode sound.wav to sound.mp2 using the default bitrate of 192 kbps 
	and using the default psychoacoustic model (model 1)</p>

<pre>
	toolame -p 2 -v 5 sound.wav newfile.mp2
</pre>
	Encode sound.wav to newfile.mp2 using psychoacoustic model 2 and encoding
	with variable bitrate. The high value of the "-v" argument means that 
	the encoding will tend to favour higher bitrates.</p>

<pre>
	toolame -p 2 -v -5 sound.wav newfile.mp2
</pre>
	Same as example above, except that the negative value of the "-v" argument
	means that the lower bitrates will be favoured over the higher ones.</p>

<pre>
	cat sound.raw | toolame -s 22050 -f -b 96 - newfile.mp2
</pre>
	Toolame is encoding from stdin at a bitrate of 96kbps and is using the
	'fast' mode which means that no psychoacoustic modelling is done.  The
        input file is raw pcm so the sample rate needs to be specified (22050Hz)</p>

<h2>Contributors</h2>

<UL>
<LI>Dist10 code writers
<LI>LAME specific contributions
<UL>
	<LI>fht routines from Ron Mayer mayer at acuson.com
	<LI>fht tweaking by Mathew Hendry math at vissci.com
	<LI>window_subband & filter_subband from LAME circa v3.30 
		(multiple LAME authors)
		(before Takehiro's window/filter/mdct combination)
</UL>
<LI> Oliver Lietz - lietz at nanocosmos.de - Tables included in the exe

<LI> Patrick de Smet - pds at telin.rug.ac.be - scale_factor calc speedup. subband_quantization speedup

<LI> Federico Grau - grauf at rfa.org -  and Bill Eldridge - bill at hk.rfa.org - option for "no padding"

<LI> Nick Burch - gagravarr at SoftHome.net - WAV file reading, os/2 Makefile mods.

<LI> Phillipe Jouguet - philippe.jouguet at vdldiffusion.com -  DAB extensions. spelling, LSF using psyII, WAVE reading [02j]

<LI> Henrik Herranen - leopold at vlsi.fi - fixed WAVE reading [02j]

<LI> Andreas Neukoetter - anti at webhome.de - verbosity patch '-t' switch for transcode plugin [02j]

<LI> Sami Sallinen - sami.sallinen at g-cluster.com - filter_subband loop unroll,  psycho_i fix for "% 1408" calcs [02j]

<LI> Ricardo Schelp - ricardoschelp at arnet.com.ar - merged window/filter subband for a nice speedup [02k]

<LI> Nicolas Croiset - ncroiset at vdl.fr - DAB length control, ignore 4GB limit when reading from stdin, fixed bitstream ending to allow concatenation of mp2 files, fixes for psycho1 model [02k]

<LI> Mike Cheng mikecheng at NOT planckenergy.com - Most of the rest 

</UL>

<h2>References</h2>

Kumar, M & Zubair, M., A high performance software implementation of mpeg audio 
encoder, 1996, ICASSP Conf Proceedings (I think)</p>

Fischer, K.A., Calculation of the psychoacoustic simultaneous masked threshold 
based on MPEG/Audio Encoder Model One, ICSI Technical Report, 1997
ftp://ftp.icsi.berkeley.edu/pub/real/kyrill/PsychoMpegOne.tar.Z </p>

Hyen-O et al, New Implementation techniques of a real-time mpeg-2 audio encoding 
system. p2287, ICASSP 99.</p>

Imai, T., et al, MPEG-1 Audio real-time encoding system, IEEE Trans on Consumer
Electronics, v44, n3 1998. p888</p>

Teh, D., et al, Efficient bit allocation algorithm for ISO/MPEG audio encoder,
Electronics Letters, v34, n8, p721</p>

Murphy, C & Anandakumar, K, Real-time MPEG-1 audio coding and decoding on a DSP
Chip, IEEE Trans on Consumer Electronics, v43, n1, 1997 p40</p>

Hans, M & Bhaskaran, V., A compliant MPEG-1 layer II audio decoder with 16-B 
arithmetic operations, IEEE Signal Proc Letters v4 n5 1997 p121</p>

[mikecheng at NOT planckenergy com] remove the NOT</p>

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